The present invention relates generally to the routing of information across networks and, more particularly, to the routing of integrated traffic across multiservice networks.
Until recently there has persisted a fundamental dichotomy between two main types of telecommunication networks. The first type of telecommunication network, the telephone network, switches and transports predominantly voice, facsimile, and modulation-demodulation system (modem) traffic. The public switched telephone network (PSTN) is an example of this type of network. Telephone networks are also deployed privately within organizations such as corporations, banks, campuses, and government offices. The second type of telecommunication network, the data network, switches or routes and transports data and video between computers. The Internet is an example of a public data network; data networks may be privately deployed.
Telephone networks were developed and deployed earlier, followed by data networks. Telephone network infrastructures are ubiquitous, however, and as a result data networks typically are built, to a limited extent, using some components of telephone networks. For example, the end user access link to a data network in some cases is implemented with a dial-up telephone line. The dial-up telephone line thus connects the end user computer equipment to the data network access gateway. Also, high speed digital trunks interconnecting remote switches and routers of a data network are often leased from telephone long-haul carriers.
Nonetheless, telephone and data network infrastructures are usually deployed together with limited sharing of resources, especially with regards to the core components of the networksxe2x80x94the switches and routers that steer the payloads throughout the networks. Furthermore, multiservice network switches are used to provide a data path, or interface, between multiple networks, each of which may operate using a different type of data or according to a different networking standard protocol. Examples of the networking protocols supported by these multiservice switches include, but are not limited to, frame relay, voice, circuit emulation, T1 channelized, E1 channelized, and Asynchronous Transfer Mode (ATM). The cost of this redundancy coupled with advances in data network technology has led, where possible, to integrated network traffic comprising voice, data, facsimile, and modem information over a unified data network. As such, a data network should now be able to accept, service, and deliver any type of data over its access links on a random, dynamic basis using a minimum set of hardware on a single platform. The problem remains, however, that a typical router or concentrator routes data through packet switch networks while voice and video traffic are routed through circuit switch networks, each of which uses different physical switch equipment.
Furthermore, in typical multiservice processing applications involving the processing of multiple data types, a matrix of digital signal processors (DSP) are typically required to perform digital signal processing operations on a number of channels of data. For modem and facsimile traffic, the DSPs are mostly used to modulate and demodulate the traffic to and from the dial-up telephone access links. For a voice call over the same link, the same DSP can instead be used to compress and decompress the voice traffic towards and from the core of the data network, to suppress undesirable echoes which usually arise at various points in the network, to suppress unnecessary silent packets to preserve network bandwidth, and to detect end-to-end voice activity to save data network bandwidth.
Within the access gateway equipment, a host bus and host processor typically communicate payload data between the DSP processors of the array and the data network side of the DSP array. While the host bus may comprise a number of channels of information, the typical system permanently assigns each channel to a particular DSP of the DSP array. Furthermore, when interfacing multiple DSPs to multiple PCM channels, a set of external logic is typically required to demultiplex and segregate each PCM channel before coupling it to an associated DSP. This approach is problematic in that it provides for inefficient allocation and use of the DSP resources, while the extra logic adversely impacts the speed and efficiency with which the information is processed.
The aforementioned desire to integrate network traffic and transport the traffic over a unified data network has heretofore resulted in a limited sharing of network resources, especially with regards to the core network switches and routers that steer the payloads throughout the networks. As such, a data network should now be able to accept, service, and deliver any type of data over its access links on a random, dynamic basis using a minimum set of hardware on a single platform. Typical routers include a group of the same dedicated hardware and software resources for each channel of information processed through the router. This scheme, however, limits the number of information channels that can be processed by a router. Furthermore, this scheme wastes router resources because, as the router accommodates many different types of data, and all of the different types of data do not require the same resources for processing, many allocated resources stand idle on the typical router. Consequently, a router is desired that provides for dynamic allocation of router resources among the received channels of information on an as-needed basis, wherein the cost, size, and complexity of the router is reduced by minimizing the duplication of resources among router channels.
The voice handling capabilities of a typical network are handled by a private branch exchange (PBX) of a public switched telephone network (PSTN). As the voice traffic becomes integrated with other types of network traffic and transported over a unified data network, however, the typical PBX becomes a limiting factor in expanding the capabilities of the unified data network. For example, the typical PBX limits voice port hunting to the PBX from which a call is initiated. Furthermore, when tie-line emulation is used to provide remote telephone extensions, the typical originating PBX does not support a call forwarding capability on ring-no-answer for the remote telephone extension. Moreover, when the remote telephone extension is not answered upon generation of a ringing signal to the extension, the typical PBX does not support disconnect supervision.
Consequently, there is a desire to expand the capabilities of the unified data network, wherein voice port hunting is performed across the network instead of being limited to the initiating or terminating private branch exchange. There exists a further desire to provide forwarding on ring-no-answer for remote telephone extensions of a unified data network. Additionally, there is a desire to provide disconnect supervision in remote telephone extensions of a unified data network.
It is therefore an object of the invention to integrate data, voice, and video onto public and private packet-based or cell-based multiservice networks comprising Frame Relay, Asynchronous Transfer Mode (ATM), High-level Data Link Control (HDLC), Internet Protocol (IP), and Time Division Multiplexed (TDM) networks, and leased line carrier services.
It is a further object of the invention to provide a trunk that is software configurable at the physical and protocol levels to support T1/E1, Frame Relay, ATM, HDLC, IP, and TDM services.
It is a further object of the invention to provide a TDM interface among a high-speed Pulse Coded Modulation (PCM) data stream and multiple processors.
It is a further object of the invention to provide dynamic allocation of multiple signal processing resources among multiple channels in voice over packet-data-network systems.
It is a further object of the invention to support voice port hunting across voice over packet-data-network systems.
It is a further object of the invention to provide forwarding on ring-no-answer for remote telephone extensions using voice over packet-data-network systems.
It is a further object of the invention to provide ringing timeout disconnect supervision in remote telephone extensions using voice over packet-data-network systems.
These and other objects of the invention are provided by a Multiservice Access Concentrator (MAC) at which a called number is received. The called number is matched against a destination-pattern index in a database. A logical group of single access telephone numbers are associated to at least one voice port, wherein a logical group of candidate voice ports are formed. The logical group of candidate voice ports are sorted to form a sorted table. The sorting comprises using a prioritized sort criteria, wherein the sort criteria comprises sorting by explicit preference, longest telephone number match metric, local voice ports taking priority over network voice ports, administrative metrics of the network voice ports, time-of-last-use timestamp, and predefined order. Hunting for a voice port among a logical group of candidate voice ports is performed, wherein the logical group of candidate voice ports is distributed across at least one voice over packet-data-network system (VOPS). The VOPS comprises voice over Internet Protocol (IP) network systems, voice over Frame Relay network systems, voice over Asynchronous Transfer Mode (ATM) network systems, and voice over High-level Data Link Control (HDLC) network systems, but the embodiment is not so limited. Ringing of a telephone extension coupled to the voice port is performed. The call is attempted to a destination-target identified by a first sorted table entry, wherein the destination-target comprises at least one local telephone interface and at least one network telephone interface. The destination-target of the network telephony interface comprises a lower layer identifier of a network endpoint and a lower layer identifier of a network circuit. The lower layer identifier of a network circuit comprises a circuit identifier of a Frame Relay circuit, a circuit identifier of an ATM circuit, a circuit identifier of an Internet Protocol (IP) address of a destination gateway, and an electronic mail address of a recipient. When the attempted call is unsuccessful, the call is reattempted to a destination target identified by a successive sorted table entry until the call attempt is successful, or the table entry is exhausted and the call is rejected.